1. Field of the Invention
The present invention relates to a multi-channel audio reproducing device and, more particularly, to a device for reproducing multi-channel audio data using two speakers and a method therefor.
2. Description of the Related Art
Endless tries to more rapidly and more exactly transmit all kinds of information, the amount of which has explosively increased in the multimedia times, result in a striking development of recent digital communication technique and in coupling of a highly integrated semiconductor (VLSI) and a signal processing technique (DSP). More still, conventionally, video, audio, and other data which have been produced and processed separately can be processed and used without a difference of information source or information media as very different formats. In this tendency, it appears that an international transmission standard of the digital data should be dispensably standardized to smoothly transmit and share the information between different types of equipment. As a result, standardization, for example, H.261 of ITU-TS in 1990, JPEG (joint picture expert group) of ISO/ITU-TS for storing and transmitting still pictures in 1992, and MPEG (moving picture expert group) of ISO/IEC was created.
Using a technique tendency of a present audio compression encoder, a wideband audio signal just like audio or music, requires much memory and a large bandwidth depending upon an increase of the volume of the data upon digitalization, storage, and transmission. To solve the above problems, many methods have been developed which are capable of encoding the audio signal, transmitting or storing the encoded signal after compression, and restoring the transmitted or stored signal as the audio signal having such an error that human beings can not recognize the same. In recent times, studies for more effectively reproducing an audio signal have being actively developed by decoding and encoding the audio signal while forming a mathematical psychoacoustic model using the auditory features of human beings. A method used for the above studies is based on the fact that in the auditory structure of human beings, the sensibility and the audible limit of recognizing a signal depending upon each frequency bandpass are different dependent upon each individual human being, and also based on the fact that the masking effect that a signal having a weaker energy than the signal having stronger energy in any frequency bandpass, can not be heard due to the signal having the stronger energy, where the signal having the weaker energy is positioned adjacent to the signal having the stronger energy. In accordance with the development of the studies of decoding and encoding all kinds of audio signals as described above, the international standardization of the ISO MPEG has been developed for the method of encoding and decoding the audio signal used in recent digital audio equipments and multimedia, the MPEG1 audio standard has been confirmed for stereo broadcasting in 1993, and the MPEG2 audio standardization has being developed at present for 5.1 channels (xe2x80x9c0.1xe2x80x9d meaning the subwoofer channel and MPEG provides a separate processing routine for the subwoofer channel). The AC3, as an independent compression algorithm of the Dolby Co. in the U.S. and centering around the recent U.S. movie industry, was determined for the high definition television (HDTV) digital audio standards of the U.S. in November, 1993, which will become one of the MPEG standard for international sharing.
These algorithms, for example, MPEG2 and AC3, play the roles of compressing the multi-channel audio data at a low transmission speed, which are adapted as the standard of the algorithm in the HDTV and DVD, so that people in a house can hear the same sound as heard in the theater. However, at least five speakers for hearing the multi-channel audio data using the above algorithm and five amps for driving these speakers are required. Actually, it is hard to include such equipment in a person""s house. Therefore, not everyone can enjoy the multi-channel audio effect therein. If the compressed multi-channel audio can be reproduced as the audio of two channels using a conventional down-mixing, the direction component of the multi-channel audio disappears, thereby providing vivid realism to listeners.
In the meanwhile, although the Dolby Pro-logic 3D-phonic algorithm invented by the Victor Co., Ltd. in Japan down-mixes the multi-channel audio signal as two channels and reproduces the down-mixed signal, it has an effect on hearing the audio as four channels.
FIG. 1 is a diagram to explain a Dolby Pro-Logic 3D-Phonic algorithm developed by the Victor Co., Ltd, in Japan. With reference to FIG. 1, reference numeral 2 indicates a processor including a Dolby Pro-Logic unit 10, and a 3D-phonic processor 12. Also, a left outputter 4 includes a left amp (LAMP) 14 and a left speaker (LSP) 16, and a right outputter 6 includes a right amp (RAMP) 18 and a right speaker (RSP) 20. Specially, FIG. 2 is a detailed circuit diagram showing the 3D-phonic processor 12 of FIG. 1.
Referring to FIGS. 1 and 2, an explanation of the operation of the algorithm will be given as follows. In FIG. 1, audio signals IL and IR of two channels to be received are changed into audio signals of four channels, that is, a left signal, a right signal, a center signal, and a surround signal (L,R,C,S) and the changed signals are applied to the 3D-phonic processor 12. In FIG. 2, regarding the operations of the 3D-phonic processor 12, the left audio signal L and the right audio signal R are respectively input to a left adder 30 and a right adder 32, the center audio signal C is commonly input to the above left and right adders 30 and 32, and the surround audio signal S is also input altogether to the above left and right adders 30 and 32 after being processed according to the 3D-phonic algorithm 34 of FIG. 2, so that the sound heard by people appears to be generated from the behind. Consequently, the left and right audio signals eL and eR including the center and surround directivity components in the left and right adders 30 and 32 are applied to the left and right lamp 14 and ramp 16, separately. Therefore, a listener can hear the audio of four channels through the left and right speakers LSP 16 and RSP 20.
However, the method of using the Dolby Pro-Logic 3D-phonic algorithm developed by the Victor Co., Ltd. in Japan has a problem in that the calculation amount is increased because the filtering for 3D-phonic and all data processing are performed only in a time domain. In addition, many signal processing devices should be equipped to quickly process the above calculation amount.
It is an object of the present invention to provide a device and a method for reproducing a multi-channel audio signal with only two speakers preserving the sound field of multi-channel audio reproduction.
It is another object of the present invention to provide a device and a method for preserving each directivity component of the multi-channel audio signal in a frequency domain.
It is a further object of the present invention to provide a device and a method for reducing the calculation amount generated when reproducing the multi-channel audio signal by using only two speakers.
The foregoing and other objects of the present invention are achieved by providing a device for reproducing multi-channel audio data to thereby provide vivid realism to a user just as multi-channel by using two speakers, including a data restorer to decode a received multi-channel audio signal and to restore the multi-channel audio data of a frequency domain; a directivity preserving processor which has a center channel direction function and a stereo surround channel direction function based on a head related transfer function indicative of the characteristic of the frequency variation due to the head of the listener for audio signals of center and stereo surround directions, to mix the center channel audio data and the stereo surround channel audio data multiplied by the direction function with left and right main channel audio data, and outputting directivity-preserved left and right main channel audio data to two main channels; and a process domain converter to convert the directivity-preserved left and right main channel audio data into the data of a time domain.